1. Field of the Invention
This invention relates to hearing aids. Further, the invention relates to a method of processing signals in a hearing aid. More specifically, it relates to a system and to a method for adapting the audio reproduction in a hearing aid to a known sound environment.
2. The Prior Art
A hearing aid usually comprises at least one microphone, a signal processing means and an output transducer, the signal processing means being adapted to receive audio signals from the microphone and to reproduce an amplified version of the input signal by the output transducer. State of the art hearing aids are programmable, relying on a programming device adapted to change the signal processing of the hearing aid to fit the hearing of a hearing aid user, i.e. to adequately amplify bands of frequencies in the user's hearing where auditive perception is deteriorated. The combination of a hearing aid and a programming device is sometimes referred to as a hearing aid system.
Hearing aids comprising means for adapting the sound reproduction to one of a plurality of different noise environments controlled either automatically or by a user according to a set of predetermined fitting rules are known, for example from U.S. Pat. No. 5,604,812, which discloses a hearing aid capable of automatic adaptation of its signal processing characteristics based on an analysis of the current ambient situation. The disclosed hearing aid comprises a signal analysis unit and a data processing unit adapted to change the signal processing characteristics of the hearing aid based on audiometric data, hearing aid characteristics and prescribable algorithms in accordance with the current acoustic environment. The specific problems of reducing background noise and improving speech intelligibility in the reproduced signal are not addressed in particular by U.S. Pat. No. 5,604,812.
In an article entitled: “Effects of fluctuating noise and interfering speech on the speech reception threshold for impaired and normal hearing”, Festen and Plomp, J. Acoust. Soc. Am, 1990, 88, pp 1725-1736, the observation is made that listeners with a sensorineural hearing loss have greater difficulty in perceiving speech masked by competing speech or modulated noise than listeners with normal hearing. The noise used is modulated in various ways, and a degree of perception is established for a representative group of both normal-hearing and hearing-impaired listeners. The difference in the perception of speech masked by unmodulated noise between listeners with normal hearing and listeners with a hearing loss is smaller than the difference in perception of speech masked by modulated noise.
A worst-case example of speech perception in modulated noise in this research is the case of noise-masking of a particular speaker with a time-reversed version of his or her own speech. In this case, the noise frequencies are similar to the speech to be perceived, and both normal-hearing listeners and hearing-impaired listeners have equal difficulties in the perception.
Thus, a need exists for a way to aid a hearing-impaired listener in perceiving and recognizing speech in modulated noise. If the character of the noise present in a given sound environment can be established with an adequate degree of certainty by a hearing aid, steps may be taken to compensate for the noise type present, and the perception of speech in that sound environment may be improved.
EP 1 129 448 B1 discloses a system and a method for measuring the signal-to-noise ratio in a speech signal. The system is capable of determining a time-dependent speech-to-noise ratio from the ratio between a time-dependent mean of the signal and a time-dependent deviation of the signal from the mean of the signal. The system utilizes a plurality of band pass filters, envelope extractors, time-local mean detectors and time-local deviation-from-mean-detectors to estimate a speech-to-noise ratio, e.g. in a hearing aid. EP 1 129 448 B1 is silent regarding speech in modulated noise.
WO 91/03042 describes a method and an apparatus for classification of a mixed speech and noise signal. The signal is split up into separate, frequency limited sub signals, each of which contains at least two harmonic frequencies of the speech signal. The envelopes of this sub signal are formed and so is a measure of synchronism between the individual envelopes of all the sub signals. The synchronism measure is compared with a threshold value for classification of the mixed signal as being significantly or insignificantly affected by the speech signal. The classification takes place with reference to an unprecedented frequency, and may therefore form the basis for a relatively precise estimate of the noise signal, in particular when this has a speech-like nature.
This method is rather complicated, as a large number of steps are required to carry out the method in practice.
Changing the audio reproduction in a hearing aid during use, for example depending on the spectral distribution of the signal processed by the hearing aid processor, might adapt the audio reproduction according to the sound of the environment to better accommodate the user's remaining hearing. An dedicated adaptation of the sound reproduction to the current sound environment may be advantageous under a lot of circumstances, for example, a different frequency response may be desired when listening to speech in quiet surroundings as compared to listening to speech in noisy surroundings. It would thus be advantageous to make the frequency response dependent on the listening situation, e.g. to provide dedicated responses for situations like a person speaking in quiet surroundings, a person speaking in noisy surroundings, or noisy surroundings without speech. In the following, the term “noise” is used to denote any unwanted signal component with respect to speech intelligibility reproduction.
Various methods for classification of listening situations suitable for use in conjunction with hearing aid systems have been devised for the purpose of identifying the prevailing type of listening situation and adapting the audio reproduction from the hearing aid to the estimated, classified listening situation. These methods may, for instance, exploit analysis of short-term RMS values at different frequencies, the modulation spectrum of the audio signal at different frequencies, or an analysis in the time domain to reveal synchronicity among different frequency bands. All these methods have shortcomings in one way or another, mainly because none of the devised methods utilize more than a mere fraction of the information available.
Another inherent problem is noise picked up from the surroundings by the hearing aid. In a modern society, the origins of the noise may often be mechanical, like transportation means, air blowers, industrial machinery or domestic appliances, or man-made, like radio or television announcements, or background chatter in a restaurant. In order for the hearing aid circuitry to be able to adapt to the noise picked up by the hearing aid, it may be advantageous to subdivide the noise environments into a plurality of different noise environment classes according to the nature and frequency distribution of the particular noise in question.
It is an object of the invention to implement strategies and methods to recognize and categorize acoustic signals from one or more hearing aid microphones and to use such information to adapt sound processing for improved user comfort. Categorization of acoustic signals implies the analysis of the current listening situation to identify which listening situation among a set of stored, specified listening situation templates the current listening situation most closely resembles. The purpose of this categorization is to select a frequency response in a hearing aid capable of producing an optimum result with respect to speech intelligibility and user comfort in the current listening situation.
A further object of the invention is to implement noise environment classification and analysis methods in a hearing aid system, making it possible to adapt sound processing to reduce the amount of noise in the reproduced signal.